[Asterisk-video] Branch to test: sdpcleanup

[Neil Stratford]

Hi all,

> Please help us test the sdpcleanup branch. Download it using svn

I've been doing some testing and have some feedback:

- You can get different behavior if debug is on or not - the ordering of 
the if/else statement starting on L4635 of chan_sip.c seems to be the 
culprit.

- The test for (numberofmediastreams > 2) on L4587 is perhaps a bit 
harsh - some clients (such as the Microsoft RTC stack) offer extra 
media, which we should ignore rather than treat as an error.

More later when I figure out why some of my calls are failing...

BTW - I have some reports of differences in behavior of the current 1.2 
branch between 1.2.4 and 1.2.8 with video codec negotiation. Has anyone 
  else seen this?

Neil

-- 
Neil Stratford - http://www.vipadia.com/ - sip:call at vipadia.com
Vipadia Limited :: VoIP Research and Development